In conventional telephony, a basic channel format is the 64 kbit/s channel, in which a single speech connection is transmitted. The 64 kbit/s channel transmits 8000 samples per seconds, each sample having 8 bits. Typically, a plurality of 64 kbit/s channels are transmitted in a single transmission line to form 1544 kbit/s, 2048 kbit/, and higher transmission rates. In a 2048 kbit/s transmission link, a 256 bit i.e. 32 byte frame is transmitted 8000 times per second. The 32 groups of 8 bits, i.e. bytes of the frame are referred to as time slots. The terms PCM transmission line and trunk line are commonly used to refer to a communication link transmitting a plurality of 64 kbit/s channels. Therefore, to identify a certain speech channel transmitted within a group of trunk lines, one needs to indicate the time slot number of the channel and a trunk line identifier. The term trunk line is sometimes used also to denote a basic 1544 kbit/s or 2048 kbit/s unit of transmission even in a case, when a plurality of such transmission units are transmitted in a single physical transmission medium i.e. in a single physical transmission line. Although the acronym PCM strictly considered denotes pulse code modulation, which is typically used in trunk lines, the term PCM transmission line is generally used by a person skilled in the art and specifically in this specification to refer to aforementioned logical group of channels or a group of groups of channels and not to a specific modulation method.
Further, packet based transmission networks are presently in widespread use, a prominent example being the Internet. A common packet transmission protocol is the Internet Protocol (IP). The IP protocol version 4 is described in detail in the specification RFC 791. The next version of the IP protocol, known as IPv6, is described in the specification RFC 1883.
The increasing importance and use of telecommunication drives toward interconnection of different types of networks. For example, the Internet is already used for transmitting voice using so called Internet telephony. As the data transmission capacity of the Internet increases, the use of Internet as a replacement of conventional telephones will become common. Some telephone operators already provide long distance calls via the Internet with a reduced rate.
Different schemes for interconnecting cellular telecommunication networks with the Internet are presently under development. The complicity of cellular telecommunication networks and the wide variety of services they provide create new and extensive fields of problems in the interconnection of different networks. One example of a feature unique to cellular telecommunication networks is the compression of speech, which is needed due to limitations of the capacity of the radio interface. A mobile station codes the speech of the user using one of the available codecs, and transmits the resulting coded speech parameters over the radio interface to the base station of the cellular network. The coded speech parameters are decoded back to a speech signal in the cellular network. However, typical compression methods used do not transmit all data in the speech signal, since the compression methods take advantage of the fact, that speech perception of a typical listener is very sensitive to certain features of a speech signal, while being less sensitive, even insensitive to some other features. Therefore, typical compression methods leave out those parts of a speech signal, which are not important to the perceived quality of transmitted speech. However, when coding and decoding is performed more than once, such as in a mobile-to-mobile connection in a cellular telecommunication network, speech quality may be drastically reduced due to the double coding and decoding. This problem can be avoided for example by using the so called tandem free operation (TFO) mode of transmission. In TFO mode, the cellular network element performing the decoding of the coded speech parameters received from the mobile station, inserts the original received coded speech parameteres into the decoded speech signal which is forwarded to the receiver. The speech parameters are typically inserted into the least significant bits of the speech samples of the speech signal, whereby they are perceived as a slight increase of background noise by a receiver of the speech signal, if the receiver does not utilize the embedded speech parameters. In case of a mobile to mobile TFO mode call, the network element at the receiving end performing the encoding of the speech signal for transmission to the receiving mobile station extracts the embedded speech parameters, and transmits those to the mobile station without performing a second coding operation. The receiving mobile station then decodes the speech parameters into a speech signal. In the TFO mode, a speech signal is coded only once, i.e. in the transmitting mobile station, and the receiving mobile station receives the coded speech parameters prepared by the transmitting mobile station, whereby double coding is avoided. This significantly improves the speech quality because without TFO, the original speech signal is coded twice with the lossy speech compression algorithm which degrades the speech quality each time the compression is applied. The difference between the single encoding and the tandem encoding becomes even more important when the bit-rate of a speech codec is very low. The old high bit-rate speech coding standards, as exemplified by the G.711 standard of 64 kbits PCM coding, are very robust to successive coding. However, the state of the art speech coders operating in a range of 4 kbits to 16 kbits are quite sensitive to more than one successive coding.
A number of problems arises when different types of transmission networks participate in transmission of connections, especially when different connections have different parameters such as the data transmission rate and whether compression is used or not. One problem, for example, is how to optimize the data transmission in the case, when some of the data transmission channels are compressed and some transmission channels are not compressed. A further problem is how to efficiently connect PCM transmission lines with an packet based network such as an IP network.